similar to: Memory Consumption

> -----Original Message----- > From: Gilad Ben-Yossef [mailto:gilad@codefidence.com] > I'm prbably stupid, but wont this do what you want? > > > exten => 1,1,Goto(bye,s,1) No, because I wanted to match on "D", not "1". Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't

Redhat 9 is running 100% cpu usage. I had a couple boxes doing this. upgraded to Fedora and its ok.

BR C.

> -----Original Message----- > From: David Brodbeck [mailto:DavidB@mail.interclean.com] > Okay, the problem appears to be that I'm tone deaf. ;) > > I finally thought to turn on debugging on the channel. The > PBX is sending > "D", not "*". The programmer of the previous voice mail system (whose > configuration I was cribbing from) seems to have

At time to time somebody is trying "their luck" and send me most likely a junk fax on my voice line. During normal working hours is not a problem I just pickup the line and hangup the call but after-hours my voice mailbox is intercepting the call and recording those "beeps" (waisting my CPU cycles). Is there a way to block call / issue hangup command if the incoming call is a

Hi everyone, This is off topic and is for GS technical support really but it seems that there are a lot of Budge Tone 100/101/102 users out there. I've got a Budge Tone-100 (101 - without the extra 10base ethernet connetion?) here. I changed the configuration through its web based interface and I clicked the reboot link. But then something went wrong and ever since then it doesn't

For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100

I have 4 gig in my * box. I'm tuning for performance and I'd like to ask opinions: 1. asterisk -p == renice -20 ?? 2. I've turned off swap with no apparent ill effects. Can anyone commment on long term effects with moderate load (say, 30 SIP phones / 2-3K calls /day) 3. Can anyone comment on using ramdisk as swap and whether this is a good idea or bad idea? I'm using 2.6

from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx "restart now" >/dev/null 2>&1 or 10 7 * * * root /usr/sbin/asterisk -r -x "restart gracefully" >/dev/null 2>&1

By default asterisk (on my system) runs with NICE value "0" like most of the programs. Will the performance improve if I assign NICE value of let say -5 or -10 to asterisk? To my understanding it should as the priority will be higher than most other programs (it might even solve some echo problem). Am I right? How to start asterisk with priority let say -5? If I start the main process

Hi, I am having trouble configuring CAPI so that call transfers work. I make a SIP call to asterisk which goes out on ISDn via CAPI. Then I try to do a transfer from the SIP phone which doesn't work and results in the call being disconnected. The error message given by asterisk is that it chan_capi can't find an entry for the outgoing msn for the transfer however the outgoing msn is the

Hello! I want to use "asterisk -rx "show version"" from a php script called in the browser using the local apache, which runs as user "apache". Asterisk is running as root. I added the following line to /etc/sudoers using visudo: apache ALL = NOPASSWD: /usr/sbin/asterisk When i am on the command line of my linux box it looks like this:

----- Forwarded message from Gilad Ben-Yossef <gilad@benyossef.com> ----- From: Gilad Ben-Yossef <gilad@benyossef.com> Organization: Codefidence ltd. A name you can trust. To: Linux-IL <linux-il@linux.org.il> Subject: [Announcement] Asterisk-IL mailing list X-Bogosity: Unsure [50.0%] X-listar-version: Listar v0.124a X-original-sender: gilad@benyossef.com X-list: linux-il

Thanks for the suggestion, C F, but the problem is there is a rather big database application behind with many users, so a static configuration is not suitable for my needs. i am working mostly with realtime and agi. regards, roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Sunday, July 03,

Hello! Is there a way to use AVM Fritz!PCI as a ZAP interface and have it configured for ZAP channels? Thanx in advance! Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in

Hello! Is it possible to run Asterisk as a SMS Service Center and does it work over a directly connected ISDN (CAPI) interface card? Did anyone already use Asterisk for that? Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

Hi to all, we are searching for a hardware based DSP solution for use with Asterisk based on PCI or MiniPCI to reduce main processor load and to use embedded boards with Digium E1/T1 cards like TE410P. does anyone know about any manufactorer of those cards or someone who is able to develop/build such cards? Specifications: PCI or MiniPCI up to 120 concurrent transcodings Codecs: G.729/G.729A or

Hi Jean-Marc, thanks for the response. First, I will clarify again that floating-point solves this - so isn't that a bug in fixed-point? Also, I understand that algorithmically the AEC won't cancel echo properly on a non-linear signal, but why completely distort the output? If the echo just won't get cancelled it would be acceptable, but in the current state it disables the ability to